@adolfoc wrote:

I’m trying to think of a way where FPBX has a cronjob that runs every day at 8pm. That cronjob would check to see if anyone is still logged into an ACD queue, and if someone is, send out an email saying which phones are logged in (or just someone is still logged in).

Basically, we used to have a procedure in our office where at the end of the day last one out the door checks to make sure everyone is logged out of the helpdesk queue. However in this WFH world we are in, harder to know who is still working at the end of the day… and if someone forgot to logout.

It’s not a huge deal because we have timeouts on the helpdesk queue for callers waiting for a while.

But I thought maybe some sort of email blast saying, “hey dude you forgot to logout” might help.

any suggestions would be helpful.

Posts: 5

Participants: 4

Read full topic

@rnrstar wrote:

This is now the second uc40 where I have had this very same issue. When I try to bring up the FreePBX gui, I get the following:
Unable to locate the FreePBX BMO Class 'Userman’A required module might be disabled or uninstalled.

I tried the recommended steps by running fwconsole ma install userman and fwconsole ma enable userman which then tells me that it’s already enabled.

I also tried fwconsole ma refresh signatures and fwconsole sa update and fwconsole chown. None of that has brought the UI back up.

This is running Asterisk 13.29.2

I really don’t want to have to do a complete reinstall of this server if at all possible. The system works in that I’m able to take and make calls. I just can’t use the UI to make any configuration changes.

Posts: 6

Participants: 3

Read full topic

@aboggs wrote:

I’m refering to FreePBX Statistics windows in Dashboard

Posts: 6

Participants: 4

Read full topic

@kfeen wrote:

Quick question. What would be the recommended way of handling multiple FreePBX systems? I have multiple clients that I’m providing a hosted PBX service, should I virtualize all of the deployments or use a separate bare metal machine for every single one?

Posts: 4

Participants: 4

Read full topic

@tim007 wrote:

This morning we discovered that our IVR has stopped responding to caller selections. I note that this was discovered after applying a few module updates (edge track); not sure if this is correlation or causation. Unfortunately, I do not remember which modules these were to roll them back.

I note that it isn’t a blanket issue with DTMF, as it works fine when I tried making selections in a VM box.

I tried temporarily removing the the IVR from the call flow and sending calls directly to one of our call queues, but now, oddly, callers are still getting the IVR message, but when it finishes, are immediately being sent to the queue.

I did an fwconsle restart to see if that would clear anything up, but it did not.

Any thoughts?

Posts: 2

Participants: 1

Read full topic

@Fabian_Luttman wrote:

Hello,
This is a new FreePBX 15.0.16.49 install from ISO.
I’ve trued to configure the Backup/Restore task, but the screen won’t let me select a Storage location. The option is grayed out…
I’m trying to post a picture of the screen. Not sure it will work:

image.png1336×109 5.84 KB

Any idea what the problem might be?
Thanks,
Fabian

Posts: 3

Participants: 2

Read full topic

@philip_rhoades wrote:

People,

I have been following Asterix from since before I retired and now I don’t have clients that might make use of a multi-user PBX but I have a few possible uses of the tech for my own non-profit organisations etc.

I installed the downloaded iso happily in a VM except for setting up a free SIP which required me to be in N America (I am in Australia).

So the first step for me is to set up a soft phone on my Fedora Linux workstation that can talk to the FPBX on the VM and just chat to someone . . somewhere else in the country or the world . . - is there another conveniet SIP provider than I can use?

Secondly, I would like to connect my single line PSTN connection to the workstation somehow - I presume this will require hardware of some sort to plug into either a USB port or a PCI slot?

Eventually I want to be able to use the setup as a digital answering machine for the PSTN line.

Thanks,
Phil.

Posts: 1

Participants: 1

Read full topic

@belac wrote:

right up front i want to let you know i know very little about phone systems. I would like to get some verification that this will work before i get into the weeds setting it up and waste a bunch of time to find out it wont work.
I am wanting to test out freepbx for my org. We have mitel phones that i would like to use. looking at the documentation it shows them as being supported. long term goal would be to migrate our existing configs from prem mitel 3300 to freepbx and decommission it but i understand it might just be used as reference for what would need to setup freepbx from scratch.

big questions, can the mitel phones connect to freepbx hosted server with out any additional equipment onsite. I purchased a sip truck from Avoxi to do some testing with and will be signing up a basic freepbx hosted server through that provider listed on the freepbx site. We have fortigate firewall, 5330 mitel phones, 1g fiber internet connection.

Posts: 6

Participants: 5

Read full topic

@edwinr wrote:

i got alot in my cdr where destination was log like ‘s [from-internal]’ instead of phone numbers or extensions.Please advise.
FreePBX 12.0.76.6

Posts: 2

Participants: 2

Read full topic

@yois wrote:

If I call a cell phone and he takes a call waiting beep, I hear my own MoH. That isn’t so bad, except in a conference call, if any user takes a “click,” the server’s MoH plays to the entire conference.

How do I stop that?

Posts: 1

Participants: 1

Read full topic

@JSBecker wrote:

PROBLEM: A manager, in another office, wants to listen as a trainee makes multiple outbound calls

Is there a way to remotely log into a Agent’s extension to hear all their calls as they make multiple calls and have multiple conversations?

Posts: 1

Participants: 1

Read full topic

@microchipmatt wrote:

My FreePBX box is 14.0.13.28. I have 500 extensions. Adding extensions was working perfectly, but today when I hit the 500th extension, a weird bug appeared. When I used to add an extension and hit submit, yes it would take a little while but no more than 30 seconds. Now that I’ve hit the 500th extension, when I hit submit, it just stays in a submission loop forever. Usually, it takes me back to the extensions list once the extension is added. I let it sit for 10 minutes, and it still stayed on submitting…(I usually use chrome)

As a test, I opened a different browser (Firefox), and the apply config button was there. So I applied the config and sure enough if I browse extensions, the new added extension is there, Yet from the browser I added it from (Chrome) it is still stuck in a submission loop never to go to the extensions page.

As another test, I submitted the extension from the gui from Firefox, SAME THING, the system sits in the submission loop…now if I open a browser I DID NOT submit the extention with, I can see the added extension, and I can apply the config from the browser I DID NOT add the extension from. This is so weird. Here are the things I’ve tried to solve the issue:

A. Cleared the cache in firefox and chome
B. Cleaed all data in firefox and chrome
C Applied updates to the FreePBX
D. Reboot FreePBX
E. Went to another computer and tried from Firefox and Chrome…same Issue.

None of the tried solutions solved the submission loop. All other options and the webpage/freepbx GUI load fast, but when I submit an extension I see this problem from any browser, and have to switch browsers to finish the extension add process…

Any Ideas?

Posts: 3

Participants: 1

Read full topic

@nickjenkey wrote:

Today I went to the admin page to modify an inbound routes settings and in the connectivity dropdown the inbound route and outbound route options are not there.

I restarted the system, ran a system update and still no menu items. The phones still work on our trunk including the priority inbound route I set up for my cellphone.

All the other menu Items in other tabs seem to be there

Any Ideas?

Posts: 4

Participants: 2

Read full topic

@kfeen wrote:

I have a FreePBX server sitting at my office behind a NAT router with ports 5060, and 10000-20000 open. There are four phones that are all remote sitting at my client’s store that are also behind a NAT router with no additional Firewall configuration.

The problem I’m having is that periodically these phones (Grandstream WP810s) are no longer ringing when receiving an incoming call, however, they can still answer the calls. This problem can be solved by rebooting the phones. Is this likely a problem with the phones or an issue with my FreePBX or firewall config?

Posts: 4

Participants: 2

Read full topic

@JJSmith wrote:

Hi, I am trying to setup a simple system(at least I hope so) using one or two phones. I am not sure if any of these scenarios will work, I have just seen similar systems on other internet talk shows but not 100% sure how they are doing it.

One phone: People can call in and leave a message saying their name & question/comment and then be placed on hold and be able to hear the show while waiting to go on air. The host can then listen to the messages during breaks and see which callers to put on air.

Two phones: people call in, talk to a screener to give name & question/comment, then get placed on hold and listen to the show while waiting to go on air.

I am not sure which one would be easier to implement. I would be okay with either scenario to be honest, I know there are professional call screening programs but they are very pricey and even then I am still not sure if they truly implement with a PBX system.

Mainly we just need a way to know what the callers want to talk about so we know which ones to put on air and which ones may be talking about a topic we’ve already moved on from, etc and just looking for the simplest way to achieve this.

Any help greatly appreciated.

Posts: 1

Participants: 1

Read full topic

@dghundt wrote:

I added a new Anveno trunk yesterday to a working call flow on my Freepbx physical box. All my other trunks are working fine.
When I dial out with Anveno instead, it appears to append a 40 to my dial number. Then I get a fast busy. CDR on Aveno site shows the appended 40.
I’ve not seen this before but I am grateful for your help.
How do I get rid of this appended number? I’m reviewing my trunk settings and do not see where a 40 could be appended.

SNGREP trace
Display Filter:
^Idx Method SIP From SIP To Msgs Source
[ ] 1 INVITE 119@192.xxx.1.xx:5060 7575937584@192.xxx.1.xx:5 6 192.xxx.1.xxx:53219
[ ] 2 INVITE 119@216.xx.xx.34 17575937584%40Anveo@sip.a 8 192.xxx.1.xxx:5060

Connected to Asterisk 14.7.5 currently running on freepbx (pid = 2219)
freepbxCLI> core set verbose 10
Console verbose was OFF and is now 10.
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x7f551ce134c0 – Strict RTP learning after remote address set to: 192.168.1.137:40024
– Executing [7575937584@from-internal:1] Macro(“SIP/119-00001cec”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/119-00001cec”, “TOUCH_MONITOR=1588992329.148406”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/119-00001cec”, “AMPUSER=119”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/119-00001cec”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/119-00001cec”, “1?Set(REALCALLERIDNUM=119)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/119-00001cec”, “AMPUSER=119”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/119-00001cec”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/119-00001cec”, “AMPUSERCIDNAME=119”) in new stack
– Executing [s@macro-user-callerid:8] ExecIf(“SIP/119-00001cec”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“SIP/119-00001cec”, “0?report”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/119-00001cec”, “AMPUSERCID=119”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/119-00001cec”, “__DIAL_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-user-callerid:12] Set(“SIP/119-00001cec”, “CALLERID(all)=“119” <119>”) in new stack
– Executing [s@macro-user-callerid:13] GotoIf(“SIP/119-00001cec”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“SIP/119-00001cec”, “1?Set(GROUP(concurrency_limit)=119)”) in new stack
– Executing [s@macro-user-callerid:15] ExecIf(“SIP/119-00001cec”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:16] NoOp(“SIP/119-00001cec”, “Macro Depth is 1”) in new stack
– Executing [s@macro-user-callerid:17] GotoIf(“SIP/119-00001cec”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [s@macro-user-callerid:18] GotoIf(“SIP/119-00001cec”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,37)
– Executing [s@macro-user-callerid:37] Set(“SIP/119-00001cec”, “CALLERID(number)=119”) in new stack
– Executing [s@macro-user-callerid:38] Set(“SIP/119-00001cec”, “CALLERID(name)=119”) in new stack
– Executing [s@macro-user-callerid:39] GotoIf(“SIP/119-00001cec”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:40] Set(“SIP/119-00001cec”, “CDR(cnam)=119”) in new stack
– Executing [s@macro-user-callerid:41] Set(“SIP/119-00001cec”, “CDR(cnum)=119”) in new stack
– Executing [s@macro-user-callerid:42] Set(“SIP/119-00001cec”, “CHANNEL(language)=en”) in new stack
– Executing [7575937584@from-internal:2] Gosub(“SIP/119-00001cec”, “sub-record-check,s,1(out,7575937584,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“SIP/119-00001cec”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“SIP/119-00001cec”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“SIP/119-00001cec”, “NOW=1588992329”) in new stack
– Executing [s@sub-record-check:4] Set(“SIP/119-00001cec”, “__DAY=08”) in new stack
– Executing [s@sub-record-check:5] Set(“SIP/119-00001cec”, “__MONTH=05”) in new stack
– Executing [s@sub-record-check:6] Set(“SIP/119-00001cec”, “__YEAR=2020”) in new stack
– Executing [s@sub-record-check:7] Set(“SIP/119-00001cec”, “__TIMESTR=20200508-224529”) in new stack
– Executing [s@sub-record-check:8] Set(“SIP/119-00001cec”, “__FROMEXTEN=119”) in new stack
– Executing [s@sub-record-check:9] Set(“SIP/119-00001cec”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“SIP/119-00001cec”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“SIP/119-00001cec”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“SIP/119-00001cec”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“SIP/119-00001cec”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“SIP/119-00001cec”, “3?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“SIP/119-00001cec”, “1?sub-record-check,out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] NoOp(“SIP/119-00001cec”, “Outbound Recording Check from 119 to 7575937584”) in new stack
– Executing [out@sub-record-check:2] Set(“SIP/119-00001cec”, “RECMODE=dontcare”) in new stack
– Executing [out@sub-record-check:3] ExecIf(“SIP/119-00001cec”, “1?Goto(routewins)”) in new stack
– Goto (sub-record-check,out,7)
– Executing [out@sub-record-check:7] Gosub(“SIP/119-00001cec”, “recordcheck,1(dontcare,out,7575937584)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“SIP/119-00001cec”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“SIP/119-00001cec”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“SIP/119-00001cec”, “”) in new stack
– Executing [out@sub-record-check:8] Return(“SIP/119-00001cec”, “”) in new stack
– Executing [7575937584@from-internal:3] ExecIf(“SIP/119-00001cec”, “0 ?Set(CDR(accountcode)=)”) in new stack
– Executing [7575937584@from-internal:4] Set(“SIP/119-00001cec”, “MOHCLASS=default”) in new stack
– Executing [7575937584@from-internal:5] Set(“SIP/119-00001cec”, “_NODEST=”) in new stack
– Executing [7575937584@from-internal:6] Macro(“SIP/119-00001cec”, “dialout-trunk,14,7575937584,off”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/119-00001cec”, “DIAL_TRUNK=14”) in new stack
– Executing [s@macro-dialout-trunk:2] ExecIf(“SIP/119-00001cec”, “0?Set(DIAL_OPTIONS=Hhtr)”) in new stack
– Executing [s@macro-dialout-trunk:3] GosubIf(“SIP/119-00001cec”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:4] ExecIf(“SIP/119-00001cec”, “0?Set(CALLERID(num)=119)”) in new stack
– Executing [s@macro-dialout-trunk:5] GotoIf(“SIP/119-00001cec”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/119-00001cec”, “DIAL_NUMBER=7575937584”) in new stack
– Executing [s@macro-dialout-trunk:7] Set(“SIP/119-00001cec”, “DIAL_TRUNK_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dialout-trunk:8] Set(“SIP/119-00001cec”, “OUTBOUND_GROUP=OUT_14”) in new stack
– Executing [s@macro-dialout-trunk:9] Set(“SIP/119-00001cec”, “DIAL_TRUNK_OPTIONS=T”) in new stack
– Executing [s@macro-dialout-trunk:10] GotoIf(“SIP/119-00001cec”, “0?nomax”) in new stack
– Executing [s@macro-dialout-trunk:11] GotoIf(“SIP/119-00001cec”, “0?chanfull”) in new stack
– Executing [s@macro-dialout-trunk:12] GotoIf(“SIP/119-00001cec”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:13] Macro(“SIP/119-00001cec”, “outbound-callerid,14”) in new stack
– Executing [s@macro-outbound-callerid:1] NoOp(“SIP/119-00001cec”, “119”) in new stack
– Executing [s@macro-outbound-callerid:2] NoOp(“SIP/119-00001cec”, “”) in new stack
– Executing [s@macro-outbound-callerid:3] NoOp(“SIP/119-00001cec”, “off”) in new stack
– Executing [s@macro-outbound-callerid:4] ExecIf(“SIP/119-00001cec”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:5] ExecIf(“SIP/119-00001cec”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:6] ExecIf(“SIP/119-00001cec”, “0?Set(REALCALLERIDNUM=119)”) in new stack
– Executing [s@macro-outbound-callerid:7] ExecIf(“SIP/119-00001cec”, “0?Set(AMPUSER=119)”) in new stack
– Executing [s@macro-outbound-callerid:8] GotoIf(“SIP/119-00001cec”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] Set(“SIP/119-00001cec”, “USEROUTCID=119”) in new stack
– Executing [s@macro-outbound-callerid:13] Set(“SIP/119-00001cec”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:14] Set(“SIP/119-00001cec”, “TRUNKOUTCID=DrBurkeyandHundt<7575648182>”) in new stack
– Executing [s@macro-outbound-callerid:15] GotoIf(“SIP/119-00001cec”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,20)
– Executing [s@macro-outbound-callerid:20] ExecIf(“SIP/119-00001cec”, “1?Set(CALLERID(all)=DrBurkeyandHundt<7575648182>)”) in new stack
– Executing [s@macro-outbound-callerid:21] ExecIf(“SIP/119-00001cec”, “1?Set(CALLERID(all)=119)”) in new stack
– Executing [s@macro-outbound-callerid:22] ExecIf(“SIP/119-00001cec”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:23] ExecIf(“SIP/119-00001cec”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:24] ExecIf(“SIP/119-00001cec”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:25] Set(“SIP/119-00001cec”, “CDR(outbound_cnum)=119”) in new stack
– Executing [s@macro-outbound-callerid:26] Set(“SIP/119-00001cec”, “CDR(outbound_cnam)=”) in new stack
– Executing [s@macro-dialout-trunk:14] GosubIf(“SIP/119-00001cec”, “1?sub-flp-14,s,1()”) in new stack
– Executing [s@sub-flp-14:1] ExecIf(“SIP/119-00001cec”, “0?Set(TARGET_FLP_14=17577575937584)”) in new stack
– Executing [s@sub-flp-14:2] GotoIf(“SIP/119-00001cec”, “0?match”) in new stack
– Executing [s@sub-flp-14:3] ExecIf(“SIP/119-00001cec”, “1?Set(TARGET_FLP_14=17575937584)”) in new stack
– Executing [s@sub-flp-14:4] GotoIf(“SIP/119-00001cec”, “1?match”) in new stack
– Goto (sub-flp-14,s,8)
– Executing [s@sub-flp-14:8] Set(“SIP/119-00001cec”, “DIAL_NUMBER=17575937584”) in new stack
– Executing [s@sub-flp-14:9] Return(“SIP/119-00001cec”, “”) in new stack
– Executing [s@macro-dialout-trunk:15] Set(“SIP/119-00001cec”, “OUTNUM=17575937584”) in new stack
– Executing [s@macro-dialout-trunk:16] Set(“SIP/119-00001cec”, “custom=SIP/Anveo”) in new stack
– Executing [s@macro-dialout-trunk:17] ExecIf(“SIP/119-00001cec”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
– Executing [s@macro-dialout-trunk:18] ExecIf(“SIP/119-00001cec”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:19] Macro(“SIP/119-00001cec”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/119-00001cec”, “”) in new stack
– Executing [s@macro-dialout-trunk:20] GotoIf(“SIP/119-00001cec”, “0?skipcrm”) in new stack
– Executing [s@macro-dialout-trunk:21] Set(“SIP/119-00001cec”, “__CRM_DIRECTION=OUTBOUND”) in new stack
– Executing [s@macro-dialout-trunk:22] Set(“SIP/119-00001cec”, “__CRM_DESTINATION=17575937584”) in new stack
– Executing [s@macro-dialout-trunk:23] Set(“SIP/119-00001cec”, “__CRM_SOURCE=119”) in new stack
– Executing [s@macro-dialout-trunk:24] AGI(“SIP/119-00001cec”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <SIP/119-00001cec>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@macro-dialout-trunk:25] Set(“SIP/119-00001cec”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
– Executing [s@macro-dialout-trunk:26] NoOp(“SIP/119-00001cec”, “CRM Finished”) in new stack
– Executing [s@macro-dialout-trunk:27] GotoIf(“SIP/119-00001cec”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:28] ExecIf(“SIP/119-00001cec”, “1?Set(CONNECTEDLINE(num,i)=17575937584)”) in new stack
– Executing [s@macro-dialout-trunk:29] ExecIf(“SIP/119-00001cec”, “1?Set(CONNECTEDLINE(name,i)=CID:119)”) in new stack
– Executing [s@macro-dialout-trunk:30] ExecIf(“SIP/119-00001cec”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)119)”) in new stack
– Executing [s@macro-dialout-trunk:31] GotoIf(“SIP/119-00001cec”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:32] Dial(“SIP/119-00001cec”, “SIP/Anveo/17575937584@Anveo,300,Tb(func-apply-sipheaders^s^1)”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– SIP/Anveo-00001ced Internal Gosub(func-apply-sipheaders,s,1) start
– Executing [s@func-apply-sipheaders:1] ExecIf(“SIP/Anveo-00001ced”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
– Executing [s@func-apply-sipheaders:2] NoOp(“SIP/Anveo-00001ced”, “Applying SIP Headers to channel”) in new stack
– Executing [s@func-apply-sipheaders:3] Set(“SIP/Anveo-00001ced”, “SIPHEADERKEYS=”) in new stack
– Executing [s@func-apply-sipheaders:4] ExecIf(“SIP/Anveo-00001ced”, “0?Set(Rheader=1)”) in new stack
– Executing [s@func-apply-sipheaders:5] While(“SIP/Anveo-00001ced”, “0”) in new stack
– Jumping to priority 9
– Executing [s@func-apply-sipheaders:10] ExecIf(“SIP/Anveo-00001ced”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
– Executing [s@func-apply-sipheaders:11] ExecIf(“SIP/Anveo-00001ced”, “0?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
– Executing [s@func-apply-sipheaders:12] Return(“SIP/Anveo-00001ced”, “”) in new stack
== Spawn extension (from-trunk, 7575937584, 1) exited non-zero on ‘SIP/Anveo-00001ced’
– SIP/Anveo-00001ced Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called SIP/Anveo/17575937584@Anveo
– SIP/Anveo-00001ced is ringing
– Got SIP response 486 “Busy Here” back from 72.9.149.69:5010
– SIP/Anveo-00001ced is busy
== Everyone is busy/congested at this time (1:1/0/0)
– Executing [s@macro-dialout-trunk:33] NoOp(“SIP/119-00001cec”, “Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17”) in new stack
– Executing [s@macro-dialout-trunk:34] GotoIf(“SIP/119-00001cec”, “0?continue,1:s-BUSY,1”) in new stack
– Goto (macro-dialout-trunk,s-BUSY,1)
– Executing [s-BUSY@macro-dialout-trunk:1] NoOp(“SIP/119-00001cec”, “Dial failed due to trunk reporting BUSY - giving up”) in new stack
– Executing [s-BUSY@macro-dialout-trunk:2] PlayTones(“SIP/119-00001cec”, “busy”) in new stack
– Executing [s-BUSY@macro-dialout-trunk:3] Busy(“SIP/119-00001cec”, “20”) in new stack
== Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on ‘SIP/119-00001cec’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 7575937584, 6) exited non-zero on ‘SIP/119-00001cec’
– Executing [h@from-internal:1] Macro(“SIP/119-00001cec”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/119-00001cec”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/119-00001cec”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“SIP/119-00001cec”, " monior file= ") in new stack
– Executing [s@macro-hangupcall:5] AGI(“SIP/119-00001cec”, “attendedtransfer-rec-restart.php,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <SIP/119-00001cec>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“SIP/119-00001cec”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/119-00001cec’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/119-00001cec’
– SIP/119-00001cec Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“SIP/119-00001cec”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“SIP/119-00001cec”, “HANGUP CAUSE: 17”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“SIP/119-00001cec”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“SIP/119-00001cec”, “MASTER CHANNEL: 1588992329.148406 = 1588992329.148406”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“SIP/119-00001cec”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“SIP/119-00001cec”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“SIP/119-00001cec”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <SIP/119-00001cec>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“SIP/119-00001cec”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/119-00001cec’
– SIP/119-00001cec Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
freepbxCLI>

Posts: 2

Participants: 1

Read full topic

@fwking4 wrote:

So i would like to see how insane I am before venturing too far down this road. We are a small CSP with about 30 customers. Most have unique requirements so we run each of those FPBX 14/Asterisk16 in a 3rd party cloud IAAS provider (Vultr). They work well until there is a noisy neighbor or a switch goes sideways.

My thought is to do as follows:
A. Secure a 1/2 rack at a very good Data Center
B. Put (2) HyperV 2016 servers with 256G + storage in a cluster (for failover); connected to a third 2016 Server running SMB3 and fiber connected to the 2 custered servers (RAID 10) - cluster is run using the recommended 3-4 NICs for data isolation.
C. Run all clients in the two servers - and as we get beyond 50 or so VMs (memory dependent) add another of the same style server to the cluster (n+1)
D. We don’t intend on running the FPBX HA module, but leverage HyperVs tools + shared SMB3 storage
E. PFsense in front of all of it w/ all Virtual IPs

Is this crazy for 100+ VMs w/ FPBX14 / Asterisk16?

Boy would give a leg for a Multi-Tenant FPBX version as you can imagine!

Posts: 3

Participants: 2

Read full topic

@freebs wrote:

Looking to get reactivated… I have moved the server once again… and no Zend resets left.
ID is 44838064

Thanks!

Posts: 1

Participants: 1

Read full topic

@VoipCool wrote:

An user has rights in the Freepbx administration to edit some extensions. Is it possible to hide the delete action so that the user cannot delete an extension?

Posts: 5

Participants: 4

Read full topic

@darkpixel wrote:

This is still a problem:

“There was an error requesting or processing your phone settings. Try refreshing the page and if this problem persists please contact your system administrator.”

I have a handful of remote phones that are refusing to communicate with my server. They are a ~3 hour drive from me. I would really love to be able to factory reset the phones through the web interface.

Any way someone at Digium can fix the firmware enough to support a factory reset?
I’m not talking about the “unsupported config changes” when using EPM–just a factory reset without having to wait until morning when staff arrive and freak out that the phones are down and then us having to walk them through factory resetting them through the phone interface.

Posts: 1

Participants: 1

Read full topic